WebRTC: Support config the bitrate of transcoding AAC to Opus. v5.0.167, v6.0.60

[chundonglinlin]

RTC support config audio bitrate by opus_bitrate or aac_bitrate.

# WebRTC streaming config for SRS.
# @see full.conf for detail config.

listen              1935;
max_connections     1000;
daemon              off;
srs_log_tank        console;

http_server {
    enabled         on;
    listen          8080;
    dir             ./objs/nginx/html;
}

http_api {
    enabled         on;
    listen          1985;
}
stats {
    network         0;
}
rtc_server {
    enabled on;
    listen 8000; # UDP port
    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
    candidate $CANDIDATE;
}

vhost __defaultVhost__ {
    rtc {
        enabled     on;
        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
        rtmp_to_rtc on;
        opus_bitrate 128000;
        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
        rtc_to_rtmp on;
        aac_bitrate 96000;
    }
    http_remux {
        enabled     on;
        mount       [vhost]/[app]/[stream].flv;
    }
}

[fairy-ju]

RTC support config audio bitrate by opus_bitrate or aac_bitrate.

# WebRTC streaming config for SRS.
# @see full.conf for detail config.

listen              1935;
max_connections     1000;
daemon              off;
srs_log_tank        console;

http_server {
    enabled         on;
    listen          8080;
    dir             ./objs/nginx/html;
}

http_api {
    enabled         on;
    listen          1985;
}
stats {
    network         0;
}
rtc_server {
    enabled on;
    listen 8000; # UDP port
    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
    candidate $CANDIDATE;
}

vhost __defaultVhost__ {
    rtc {
        enabled     on;
        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
        rtmp_to_rtc on;
        opus_bitrate 128000;
        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
        rtc_to_rtmp on;
        aac_bitrate 96000;
    }
    http_remux {
        enabled     on;
        mount       [vhost]/[app]/[stream].flv;
    }
}

@chundonglinlin您好，我删除并使用git clone -b develop https://gitee.com/ossrs/srs.git 重装了最新的dev版本，但发现不管是rtmp到WebRTC还是srt到WebRTC，音频码率都还是保持默认值不随配置文件而变化，我配置文件里面写的192000，但实际只有大概64000，能帮忙看看问题所在吗，谢谢！下面是我的配置文件：

Hello, I have deleted and reinstalled the latest dev version using git clone -b develop https://gitee.com/ossrs/srs.git. However, I found that regardless of whether it is RTMP to WebRTC or SRT to WebRTC, the audio bitrate still remains the default value and does not change with the configuration file. I wrote 192000 in my configuration file, but in reality, it's only about 64000. Could you help me see what the problem is? Thank you! Below is my configuration file:

listen              1935;
max_connections     1000;
daemon              off;
srs_log_tank        console;

http_api {
    enabled         on;
    listen          1985;
    https {
        # Whether enable HTTPS API.
        # default: off
        enabled on;
        # The listen endpoint for HTTPS API.
        # default: 1990
        listen 1990;
        # The SSL private key file, generated by:
        #       openssl genrsa -out server.key 2048
        # default: ./conf/server.key
        key ./conf/server.key;
        # The SSL public cert file, generated by:
        #       openssl req -new -x509 -key server.key -out server.crt -days 3650 -subj "/C=CN/ST=Beijing/L=Beijing/O=Me/OU=Me/CN=ossrs.net"
        # default: ./conf/server.crt
        cert ./conf/server.crt;
    }
}

http_server {
    enabled         on;
    listen          8080;
    dir             ./objs/nginx/html;
}

srt_server {
    enabled on;
    listen 10080;
    maxbw 1000000000;
    connect_timeout 4000;
    peerlatency 0;
    recvlatency 0;
}

rtc_server {
    enabled on;
    # Listen at udp://8000
    listen 8000;
    #
    # The $CANDIDATE means fetch from env, if not configed, use * as default.
    #
    # The * means retrieving server IP automatically, from all network interfaces,
    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
    #candidate $CANDIDATE;
    candidate *; #支持任意IP，就意味着让服务器自己选择，先选公网IP，然后选内网IP
    use_auto_detect_network_ip on; #若关闭这个功能，则不会自动选择IP
    ip_family ipv4; #自动选择IP时，选择IPv4还是IPv6的地址
    api_as_candidates on; #是否开启这个功能。若API是单独的服务器，可以关闭这个功能
    resolve_api_domain on; #若API是域名，是否将域名解析为IP地址。注意Firefox不支持域名，所以一般是推荐打开的
    keep_api_domain on;  #是否保留API的域名，支持域名解析的客户端可以自己解析IP地址，避免服务器实现解析
}

# @doc https://github.com/ossrs/srs/issues/1147#issuecomment-577607026
vhost __defaultVhost__ {
    srt {
        enabled     on;
    }
    rtc {
        enabled     on;
        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
        rtmp_to_rtc on;
		opus_bitrate 192000;
        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
        rtc_to_rtmp on;
		aac_bitrate 192000;
    }
    http_remux {
        enabled     on;
        mount       [vhost]/[app]/[stream].flv;
    }
}